I/O: WORDS
OUTPUT
1644: 16-bit/44.1 kHz PCM audio.
2444: 24-bit/44.1 kHz PCM audio.
2448: 24-bit/48 kHz PCM audio.
2488: 24-bit/88.2 kHz PCM audio.
2496: 24-bit/96 kHz PCM audio.
24176: 24-bit/176.4 kHz PCM audio.
24192: 24-bit/192 kHz PCM audio.
256 AAC: 256 kbps/44.1 kHz AAC constrained variable bit rate (CVBR) audio.
320 MP3: 320 kbps/44.1 kHz MP3 constant bit rate (CBR) audio.
3244: 32-bit (float)/44.1 kHz PCM audio.
3248: 32-bit (float)/48 kHz PCM audio.
3288: 32-bit (float)/88.2 kHz PCM audio.
3296: 32-bit (float)/96 kHz PCM audio.
32176: 32-bit (float)/176.4 kHz PCM audio.
32192: 32-bit (float)/192 kHz PCM audio.
24, 32, 48, 64, 96, 128, 160, 192, 256, 320: Common lossy file format bit rates (measured in kbps). A sample rate of 44.1 kHz is assumed, unless otherwise specified.
-A-
AAC: Advanced Audio Coding. A lossy coding format for digital audio. AAC was originally conceived as the successor to MP3. It is the default file format for iTunes and Apple Music, utilizing the iTunes Plus codec with a constrained variable bit rate of 256 kbps. The file extension is .m4a.
AIFF: Audio Interchange File Format. An uncompressed linear PCM file format for storing digital audio data. AIFF was developed by Apple. The sound quality is identical to that of WAV files. The file extension is .aif.
ALAC: Apple Lossless Audio Codec. Also known as Apple Lossless. A lossless coding format for digital audio developed by Apple. ALAC uses lossless data compression to reduce the file size to between 40 and 60 percent of the original size. The bit rate is reduced as well. Unlike lossy formats, ALAC decompresses to an identical copy of the original audio file (WAV, AIFF). In addition, ALAC supports the encoding of high resolution/high sample rate PCM audio files. The file extension is .m4a, referring only to the file’s container and not the audio encoding itself.
-B-
Bit Depth: Also known as wordlength. The number of bits required to define a PCM audio sample. In integer (fixed point) formats, the bit depth determines the resolution of the recording. Each bit represents 6 dB (technically 6.02 dB) of dynamic range. Thus, a 24-bit recording has a theoretical dynamic range of 144 dB.
Bit Rate: The number of bits processed over a given time period. In digital audio, the bit rate is measured in kilobits per second (kbps). Resolution normally increases as the bit rate becomes higher.
-C-
CBR: Constant Bit Rate. A lossy encoding method where the bit rate remains the same throughout the encoding process.
CD: Compact Disc. Also known as CD-DA (Compact Disc-Digital Audio). An optical disc format which stores 16-bit/44.1 kHz PCM audio. The CD was developed jointly by Sony and Philips and released in 1982.
CD-Text: Metadata which is embedded in the subcode of the CD providing artist, title, and other information.
Codec: A device or software application for encoding and decoding a digital audio stream.
-D-
DAC: Digital-to-Analog Converter. Also known as D/A. A device that converts a digital audio signal into an analog signal.
DAW: Digital Audio Workstation. A computer-based workstation used for recording, editing, and outputting digital audio files. In addition to a computer, a DAW usually consists of an audio interface and digital editing software.
dB: Decibel. A relative unit of loudness measurement equivalent to the Loudness Unit (LU). 1 dB = 1 LU.
dBFS: Decibels relative to full scale. An absolute unit of loudness measurement equivalent to Loudness Units relative to full scale (LUFS). -1 dBFS = -1 LUFS.
dBTP: dB True Peak referenced to 0 dBFS. True Peaks may exceed 0 dBFS.
DDP: Disc Description Protocol. A proprietary format developed by Doug Carson and Associates (DCA, Inc.) for specifying the content of optical discs, including CDs and DVDs. DDP is commonly used for delivery of a CD pre-master to a CD replicator or duplicator. A DDP pre-master consists of a ZIP file containing a folder with the DDP file set.
Decoder: In digital audio, a device or software application which converts a compressed audio stream back into an uncompressed linear PCM audio stream.
Dither: Also known as wordlength reduction (WLR). A form of noise which, when applied prior to rounding or truncation, randomizes quantization errors. Quantization errors occur whenever converting digital audio from a higher bit depth to a lower one.
Double Sample Rate: PCM audio with a sample rate of 88.2 or 96 kHz.
DSD: Direct Stream Digital. An encoding method, technically pulse-density modulation (PDM), used to digitally represent an analog audio signal. During encoding, the analog signal is stored as sigma-delta modulated digital audio, a sequence of 1-bit values at an ultra high sample rate.
DSD64: A 1-bit digital audio format with a sample rate of 2.8224 MHz (64 times that of the 44.1 kHz CD sample rate). DSD64 was originally conceived for the Sony-Phillips Super Audio CD (SACD) format.
DSD128: Also known as Double-rate DSD. A 1-bit digital audio format with a sample rate of 5.6448 MHz (128 times that of the 44.1 kHz CD sample rate).
DSF: DSD Storage Facility. A PDM file format for storing DSD audio. The file extension is .dsf.
DXD: Digital eXtreme Definition. A 32-bit (float)/352.8 kHz PCM format adopted by Merging Technologies to perform editing and processing of DSD audio.
-E-
Encoder: In digital audio, a device or software application which converts an uncompressed linear PCM audio stream into a compressed audio stream.
-F-
FLAC: Free Lossless Audio Codec. An open source lossless coding format for digital audio. FLAC uses lossless data compression to reduce the file size to between 50 and 70 percent of the original size. The bit rate is reduced as well. Unlike lossy formats, FLAC decompresses to an identical copy of the original audio file (WAV, AIFF). In addition, FLAC supports the encoding of high resolution/high sample rate PCM audio files. The file extension is .flac.
-H-
High Resolution: PCM audio with a bit depth greater than 16 bits.
High Sample Rate: PCM audio with a sample rate of 88.2, 96, 176.4, or 192 kHz.
-I-
I: Integrated. The overall program loudness measurement between start and end points (measured in LU or LUFS).
ID3 Tag: A metadata container stored within an MP3, AAC, or other digital audio file providing artist, title, and other information.
ISRC: International Standard Recording Code. The international identification system for sound and music video recordings. Each ISRC is a unique and permanent identifier for a specific recording, independent of the format on which it appears.
-L-
Lossless Coding: Lossless file formats (FLAC, ALAC, etc.) utilize linear prediction and run-length encoding to reduce the file size to between 40 and 70 percent of the original size. The bit rate is reduced as well. Unlike lossy formats, lossless formats decompress to an identical copy of the original audio file (WAV, AIFF).
Lossy Coding: Most lossy file formats (MP3, AAC, etc.) utilize perceptual coding to discard audio components that are considered beyond human perception. In effect, the audio signal is encoded using inexact approximations. The lossy data compression substantially reduces the bit rate and, therefore, file size when compared to the original uncompressed file (WAV, AIFF). Because of the reduced bit rate and file size, lossy formats are an efficient means for digital distribution and streaming.
LN: Loudness Normalization. A system for adjusting the gain of audio files with the goal of each file being reproduced at the same perceived loudness.
LRA: Loudness Range. The overall program range from the softest point to the loudest (measured in LU). The top 5% and the lowest 10% of the total loudness range are excluded from the LRA measurement to avoid extreme events from affecting the overall result.
LU: Loudness Unit. A relative unit of loudness measurement equivalent to the decibel (dB). 1 LU = 1 dB.
LUFS: Loudness Units relative to full scale. An absolute unit of loudness measurement equivalent to decibels relative to full scale (dBFS). -1 LUFS = -1 dBFS.
-M-
M: Momentary. The summed momentary loudness measurement using an integration time of 400 ms (measured in LU or LUFS).
Metadata: A set of data which describes information about other data.
Mm: Mmax. The maximum momentary loudness measurement throughout the program (measured in LU or LUFS).
MP3: MPEG -1 Audio Layer III. A lossy coding format for digital audio. MP3 is a truly universal file format which can be played back by virtually any media player. Codecs include Fraunhofer and LAME. The file extension is .mp3.
-O-
Ogg Vorbis: An open source lossy coding format for digital audio. Similar to MP3 and AAC, Ogg Vorbis uses lossy data compression to substantially reduce the bit rate and, therefore, file size when compared to the original uncompressed file (WAV, AIFF). Technically, Vorbis is the name of the codec, while OGG denotes the file container type. The file extension is .ogg.
-P-
PCM: Pulse-code Modulation. An encoding method used to digitally represent an analog audio signal. During encoding, the amplitude of the analog signal is sampled at uniform intervals and each step is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a type of PCM where the quantization levels are linearly uniform.
PDM: Pulse-density Modulation. An encoding method used to digitally represent an analog audio signal. During encoding, the analog signal is stored as sigma-delta modulated digital audio, a sequence of 1-bit values at an ultra high sample rate. PDM is used for DSD audio formats.
PMCD: Pre-master CD. A physical CD-R used for delivery of a CD pre-master to a CD duplicator.
PQ Sheet: A track list containing CD markers for track start, pregap, and subindex, in addition to a listing of the total number of tracks and the total playing time of the CD. CD-Text, UPC/EAN, and ISRC information are usually included on the PQ sheet as well. The term is also used when referring to a vinyl pre-master track list.
-Q-
QC: Quality Control.
Quad Sample Rate: PCM audio with a sample rate of 176.4 or 192 kHz.
-R-
ReplayGain: An open source loudness normalization (LN) system used in some media players and on Spotify which measures the peak level and perceived loudness of an audio file and attempts to adjust the average loudness to a target level of -18 LUFS (ReplayGain 2.0). A peak level of 1.000 is equivalent to 0 dBFS.
-S-
S: Short-term. The summed short–term loudness measurement using an integration time of 3 seconds and a sliding window (measured in LU or LUFS).
Sample Peak: The highest sample level in a digital audio file.
Sample Rate: In PCM audio, the number of samples carried per second (measured in Hz or kHz).
Sample Rate Conversion: Also known as resampling. The process of changing the sample rate of digital audio to a new sample rate by calculating the values of the new samples directly from the current samples. Upsampling is a sample rate conversion to a higher sample rate, whereas downsampling is to a lower sample rate.
SDM: Sigma-delta Modulation. An encoding method used to digitally represent an analog audio signal. In delta modulation, the change in the signal, rather than the absolute value, is encoded. The result of this process is a bitstream, as opposed to a stream of coded samples as with PCM.
Single Sample Rate: PCM audio with a sample rate of 44.1 or 48 kHz.
Sm: Smax. The maximum short-term loudness measurement throughout the program (measured in LU or LUFS).
Sound Check: Apple’s proprietary loudness normalization (LN) system used in iTunes and on Apple Music which measures the peak level and perceived loudness of an audio file and attempts to adjust the average loudness to a target level of approximately -16 LUFS.
-T-
Target Level: The level to which a loudness normalization system attempts to normalize (normally measured in LUFS).
TP: True Peak. Also known as Intersample Peak (ISP). A peak measurement identifying potential clipping generated in large part during digital to analog conversion, sample rate conversion, and lossy file encoding/decoding (measured in dBTP). A True Peak meter measures sample peaks and estimates ISPs. A standard sample peak meter does not estimate ISPs. True Peaks may exceed 0 dBFS.
TPm: TPmax. The maximum True Peak measurement throughout the program (measured in dBTP).
-U-
UPC/EAN: Universal Product Code/European Article Number. Also known as Media Catalog Number (MCN). The standard product code associated with a retail product.
-V-
VBR: Variable Bit Rate. A lossy encoding method where the bit rate varies according to musical complexity during the encoding process. Constrained Variable Bit Rate (CVBR) limits the average bit rate variation by specifying a lower bit rate limit.
-W-
WAV: Waveform Audio File Format. Also known as WAVE. An uncompressed linear PCM file format for storing digital audio data. WAV was developed jointly by Microsoft and IBM. The sound quality is identical to that of AIFF files. The file extension is .wav.